add external sip and rtp along with enum dialplan to default so ISN and enum work out of the box

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@5930 d0543943-73ff-0310-b7d9-9358b9ac24b2

conf/sofia.conf.xml

@@ -44,7 +44,7 @@
 	<param name="debug" value="1"/>
 	<param name="rfc2833-pt" value="101"/>
 	<param name="sip-port" value="5060"/>
-	<param name="dialplan" value="XML"/>
+	<param name="dialplan" value="enum,XML"/>
 	<param name="dtmf-duration" value="100"/>
 	<param name="codec-prefs" value="$${global_codec_prefs}"/>
 	<param name="codec-ms" value="20"/>
@@ -73,7 +73,7 @@
 
 	<!-- this lets anything register -->
 	<!--  comment the next line and uncomment one or both of the other 2 lines for call authentication -->
-	<param name="accept-blind-reg" value="true"/>
+	<!-- <param name="accept-blind-reg" value="true"/> -->
 
 	<!--TTL for nonce in sip auth-->
 	<param name="nonce-ttl" value="60"/>
@@ -84,7 +84,8 @@
 	<!-- on authed calls, authenticate *all* the packets not just invite -->
 	<!--<param name="auth-all-packets" value="true"/>-->
 
-	<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
+	<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
+	<param name="ext-rtp-ip" value="$${external_sip_ip}"/>
 
 	<!-- <param name="ext-sip-ip" value="100.101.102.103"/> -->
 	<!-- VAD choose one (out is a good choice); -->